Merge 5.15-rc6 into tty-next

We need the serial/tty fixes in here as well.

Signed-off-by: Greg Kroah-Hartman <gregkh@linuxfoundation.org>

38 files changed, 506 insertions, 136 deletions

diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c
index a59de24695ec..dfe5a64e19d2 100644
--- a/sound/core/pcm_compat.c
+++ b/sound/core/pcm_compat.c
@@ -468,6 +468,76 @@ static int snd_pcm_ioctl_sync_ptr_x32(struct snd_pcm_substream *substream,
 }
 #endif /* CONFIG_X86_X32 */
 
+#ifdef __BIG_ENDIAN
+typedef char __pad_before_u32[4];
+typedef char __pad_after_u32[0];
+#else
+typedef char __pad_before_u32[0];
+typedef char __pad_after_u32[4];
+#endif
+
+/* PCM 2.0.15 API definition had a bug in mmap control; it puts the avail_min
+ * at the wrong offset due to a typo in padding type.
+ * The bug hits only 32bit.
+ * A workaround for incorrect read/write is needed only in 32bit compat mode.
+ */
+struct __snd_pcm_mmap_control64_buggy {
+	__pad_before_u32 __pad1;
+	__u32 appl_ptr;
+	__pad_before_u32 __pad2;	/* SiC! here is the bug */
+	__pad_before_u32 __pad3;
+	__u32 avail_min;
+	__pad_after_uframe __pad4;
+};
+
+static int snd_pcm_ioctl_sync_ptr_buggy(struct snd_pcm_substream *substream,
+					struct snd_pcm_sync_ptr __user *_sync_ptr)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_pcm_sync_ptr sync_ptr;
+	struct __snd_pcm_mmap_control64_buggy *sync_cp;
+	volatile struct snd_pcm_mmap_status *status;
+	volatile struct snd_pcm_mmap_control *control;
+	int err;
+
+	memset(&sync_ptr, 0, sizeof(sync_ptr));
+	sync_cp = (struct __snd_pcm_mmap_control64_buggy *)&sync_ptr.c.control;
+	if (get_user(sync_ptr.flags, (unsigned __user *)&(_sync_ptr->flags)))
+		return -EFAULT;
+	if (copy_from_user(sync_cp, &(_sync_ptr->c.control), sizeof(*sync_cp)))
+		return -EFAULT;
+	status = runtime->status;
+	control = runtime->control;
+	if (sync_ptr.flags & SNDRV_PCM_SYNC_PTR_HWSYNC) {
+		err = snd_pcm_hwsync(substream);
+		if (err < 0)
+			return err;
+	}
+	snd_pcm_stream_lock_irq(substream);
+	if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_APPL)) {
+		err = pcm_lib_apply_appl_ptr(substream, sync_cp->appl_ptr);
+		if (err < 0) {
+			snd_pcm_stream_unlock_irq(substream);
+			return err;
+		}
+	} else {
+		sync_cp->appl_ptr = control->appl_ptr;
+	}
+	if (!(sync_ptr.flags & SNDRV_PCM_SYNC_PTR_AVAIL_MIN))
+		control->avail_min = sync_cp->avail_min;
+	else
+		sync_cp->avail_min = control->avail_min;
+	sync_ptr.s.status.state = status->state;
+	sync_ptr.s.status.hw_ptr = status->hw_ptr;
+	sync_ptr.s.status.tstamp = status->tstamp;
+	sync_ptr.s.status.suspended_state = status->suspended_state;
+	sync_ptr.s.status.audio_tstamp = status->audio_tstamp;
+	snd_pcm_stream_unlock_irq(substream);
+	if (copy_to_user(_sync_ptr, &sync_ptr, sizeof(sync_ptr)))
+		return -EFAULT;
+	return 0;
+}
+
 /*
  */
 enum {
@@ -537,7 +607,7 @@ static long snd_pcm_ioctl_compat(struct file *file, unsigned int cmd, unsigned l
 		if (in_x32_syscall())
 			return snd_pcm_ioctl_sync_ptr_x32(substream, argp);
 #endif /* CONFIG_X86_X32 */
-		return snd_pcm_common_ioctl(file, substream, cmd, argp);
+		return snd_pcm_ioctl_sync_ptr_buggy(substream, argp);
 	case SNDRV_PCM_IOCTL_HW_REFINE32:
 		return snd_pcm_ioctl_hw_params_compat(substream, 1, argp);
 	case SNDRV_PCM_IOCTL_HW_PARAMS32:
diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c
index 6c0a4a67ad2e..6f30231bdb88 100644
--- a/sound/core/rawmidi.c
+++ b/sound/core/rawmidi.c
@@ -873,12 +873,21 @@ static long snd_rawmidi_ioctl(struct file *file, unsigned int cmd, unsigned long
 			return -EINVAL;
 		}
 	}
+	case SNDRV_RAWMIDI_IOCTL_USER_PVERSION:
+		if (get_user(rfile->user_pversion, (unsigned int __user *)arg))
+			return -EFAULT;
+		return 0;
+
 	case SNDRV_RAWMIDI_IOCTL_PARAMS:
 	{
 		struct snd_rawmidi_params params;
 
 		if (copy_from_user(&params, argp, sizeof(struct snd_rawmidi_params)))
 			return -EFAULT;
+		if (rfile->user_pversion < SNDRV_PROTOCOL_VERSION(2, 0, 2)) {
+			params.mode = 0;
+			memset(params.reserved, 0, sizeof(params.reserved));
+		}
 		switch (params.stream) {
 		case SNDRV_RAWMIDI_STREAM_OUTPUT:
 			if (rfile->output == NULL)
diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c
index 382275c5b193..7f3fd8eb016f 100644
--- a/sound/core/seq_device.c
+++ b/sound/core/seq_device.c
@@ -156,6 +156,8 @@ static int snd_seq_device_dev_free(struct snd_device *device)
 	struct snd_seq_device *dev = device->device_data;
 
 	cancel_autoload_drivers();
+	if (dev->private_free)
+		dev->private_free(dev);
 	put_device(&dev->dev);
 	return 0;
 }
@@ -183,11 +185,7 @@ static int snd_seq_device_dev_disconnect(struct snd_device *device)
 
 static void snd_seq_dev_release(struct device *dev)
 {
-	struct snd_seq_device *sdev = to_seq_dev(dev);
-
-	if (sdev->private_free)
-		sdev->private_free(sdev);
-	kfree(sdev);
+	kfree(to_seq_dev(dev));
 }
 
 /*
diff --git a/sound/drivers/pcsp/pcsp_lib.c b/sound/drivers/pcsp/pcsp_lib.c
index ed40d0f7432c..773db4bf0876 100644
--- a/sound/drivers/pcsp/pcsp_lib.c
+++ b/sound/drivers/pcsp/pcsp_lib.c
@@ -143,7 +143,7 @@ enum hrtimer_restart pcsp_do_timer(struct hrtimer *handle)
 	if (pointer_update)
 		pcsp_pointer_update(chip);
 
-	hrtimer_forward(handle, hrtimer_get_expires(handle), ns_to_ktime(ns));
+	hrtimer_forward_now(handle, ns_to_ktime(ns));
 
 	return HRTIMER_RESTART;
 }
diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c
index 5388b85fb60e..a18c2c033e83 100644
--- a/sound/firewire/motu/amdtp-motu.c
+++ b/sound/firewire/motu/amdtp-motu.c
@@ -276,10 +276,11 @@ static void __maybe_unused copy_message(u64 *frames, __be32 *buffer,
 
 	/* This is just for v2/v3 protocol. */
 	for (i = 0; i < data_blocks; ++i) {
-		*frames = (be32_to_cpu(buffer[1]) << 16) |
-			  (be32_to_cpu(buffer[2]) >> 16);
+		*frames = be32_to_cpu(buffer[1]);
+		*frames <<= 16;
+		*frames |= be32_to_cpu(buffer[2]) >> 16;
+		++frames;
 		buffer += data_block_quadlets;
-		frames++;
 	}
 }
 
diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c
index cb5b5e3a481b..daf731364695 100644
--- a/sound/firewire/oxfw/oxfw.c
+++ b/sound/firewire/oxfw/oxfw.c
@@ -184,13 +184,16 @@ static int detect_quirks(struct snd_oxfw *oxfw, const struct ieee1394_device_id
 			model = val;
 	}
 
-	/*
-	 * Mackie Onyx Satellite with base station has a quirk to report a wrong
-	 * value in 'dbs' field of CIP header against its format information.
-	 */
-	if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE)
+	if (vendor == VENDOR_LOUD) {
+		// Mackie Onyx Satellite with base station has a quirk to report a wrong
+		// value in 'dbs' field of CIP header against its format information.
 		oxfw->quirks |= SND_OXFW_QUIRK_WRONG_DBS;
 
+		// OXFW971-based models may transfer events by blocking method.
+		if (!(oxfw->quirks & SND_OXFW_QUIRK_JUMBO_PAYLOAD))
+			oxfw->quirks |= SND_OXFW_QUIRK_BLOCKING_TRANSMISSION;
+	}
+
 	return 0;
 }
 
diff --git a/sound/hda/hdac_controller.c b/sound/hda/hdac_controller.c
index 062da7a7a586..f7bd6e2db085 100644
--- a/sound/hda/hdac_controller.c
+++ b/sound/hda/hdac_controller.c
@@ -421,8 +421,9 @@ int snd_hdac_bus_reset_link(struct hdac_bus *bus, bool full_reset)
 	if (!full_reset)
 		goto skip_reset;
 
-	/* clear STATESTS */
-	snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK);
+	/* clear STATESTS if not in reset */
+	if (snd_hdac_chip_readb(bus, GCTL) & AZX_GCTL_RESET)
+		snd_hdac_chip_writew(bus, STATESTS, STATESTS_INT_MASK);
 
 	/* reset controller */
 	snd_hdac_bus_enter_link_reset(bus);
diff --git a/sound/pci/hda/hda_bind.c b/sound/pci/hda/hda_bind.c
index 2523b23389e9..1c8bffc3eec6 100644
--- a/sound/pci/hda/hda_bind.c
+++ b/sound/pci/hda/hda_bind.c
@@ -298,29 +298,31 @@ int snd_hda_codec_configure(struct hda_codec *codec)
 {
 	int err;
 
+	if (codec->configured)
+		return 0;
+
 	if (is_generic_config(codec))
 		codec->probe_id = HDA_CODEC_ID_GENERIC;
 	else
 		codec->probe_id = 0;
 
-	err = snd_hdac_device_register(&codec->core);
-	if (err < 0)
-		return err;
+	if (!device_is_registered(&codec->core.dev)) {
+		err = snd_hdac_device_register(&codec->core);
+		if (err < 0)
+			return err;
+	}
 
 	if (!codec->preset)
 		codec_bind_module(codec);
 	if (!codec->preset) {
 		err = codec_bind_generic(codec);
 		if (err < 0) {
-			codec_err(codec, "Unable to bind the codec\n");
-			goto error;
+			codec_dbg(codec, "Unable to bind the codec\n");
+			return err;
 		}
 	}
 
+	codec->configured = 1;
 	return 0;
-
- error:
-	snd_hdac_device_unregister(&codec->core);
-	return err;
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_configure);
diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c
index a9ebefd60cf6..0c4a337c9fc0 100644
--- a/sound/pci/hda/hda_codec.c
+++ b/sound/pci/hda/hda_codec.c
@@ -791,6 +791,7 @@ void snd_hda_codec_cleanup_for_unbind(struct hda_codec *codec)
 	snd_array_free(&codec->nids);
 	remove_conn_list(codec);
 	snd_hdac_regmap_exit(&codec->core);
+	codec->configured = 0;
 }
 EXPORT_SYMBOL_GPL(snd_hda_codec_cleanup_for_unbind);
 
diff --git a/sound/pci/hda/hda_controller.c b/sound/pci/hda/hda_controller.c
index 7cd452831fd3..930ae4002a81 100644
--- a/sound/pci/hda/hda_controller.c
+++ b/sound/pci/hda/hda_controller.c
@@ -25,6 +25,7 @@
 #include <sound/core.h>
 #include <sound/initval.h>
 #include "hda_controller.h"
+#include "hda_local.h"
 
 #define CREATE_TRACE_POINTS
 #include "hda_controller_trace.h"
@@ -1248,17 +1249,24 @@ EXPORT_SYMBOL_GPL(azx_probe_codecs);
 int azx_codec_configure(struct azx *chip)
 {
 	struct hda_codec *codec, *next;
+	int success = 0;
 
-	/* use _safe version here since snd_hda_codec_configure() deregisters
-	 * the device upon error and deletes itself from the bus list.
-	 */
-	list_for_each_codec_safe(codec, next, &chip->bus) {
-		snd_hda_codec_configure(codec);
+	list_for_each_codec(codec, &chip->bus) {
+		if (!snd_hda_codec_configure(codec))
+			success++;
 	}
 
-	if (!azx_bus(chip)->num_codecs)
-		return -ENODEV;
-	return 0;
+	if (success) {
+		/* unregister failed codecs if any codec has been probed */
+		list_for_each_codec_safe(codec, next, &chip->bus) {
+			if (!codec->configured) {
+				codec_err(codec, "Unable to configure, disabling\n");
+				snd_hdac_device_unregister(&codec->core);
+			}
+		}
+	}
+
+	return success ? 0 : -ENODEV;
 }
 EXPORT_SYMBOL_GPL(azx_codec_configure);
 
diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h
index 3062f87380b1..f5bf295eb830 100644
--- a/sound/pci/hda/hda_controller.h
+++ b/sound/pci/hda/hda_controller.h
@@ -41,7 +41,7 @@
 /* 24 unused */
 #define AZX_DCAPS_COUNT_LPIB_DELAY  (1 << 25)	/* Take LPIB as delay */
 #define AZX_DCAPS_PM_RUNTIME	(1 << 26)	/* runtime PM support */
-/* 27 unused */
+#define AZX_DCAPS_RETRY_PROBE	(1 << 27)	/* retry probe if no codec is configured */
 #define AZX_DCAPS_CORBRP_SELF_CLEAR (1 << 28)	/* CORBRP clears itself after reset */
 #define AZX_DCAPS_NO_MSI64      (1 << 29)	/* Stick to 32-bit MSIs */
 #define AZX_DCAPS_SEPARATE_STREAM_TAG	(1 << 30) /* capture and playback use separate stream tag */
diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c
index 3aa432d814a2..4d22e7adeee8 100644
--- a/sound/pci/hda/hda_intel.c
+++ b/sound/pci/hda/hda_intel.c
@@ -307,7 +307,8 @@ enum {
 /* quirks for AMD SB */
 #define AZX_DCAPS_PRESET_AMD_SB \
 	(AZX_DCAPS_NO_TCSEL | AZX_DCAPS_AMD_WORKAROUND |\
-	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME)
+	 AZX_DCAPS_SNOOP_TYPE(ATI) | AZX_DCAPS_PM_RUNTIME |\
+	 AZX_DCAPS_RETRY_PROBE)
 
 /* quirks for Nvidia */
 #define AZX_DCAPS_PRESET_NVIDIA \
@@ -883,10 +884,11 @@ static unsigned int azx_get_pos_skl(struct azx *chip, struct azx_dev *azx_dev)
 	return azx_get_pos_posbuf(chip, azx_dev);
 }
 
-static void azx_shutdown_chip(struct azx *chip)
+static void __azx_shutdown_chip(struct azx *chip, bool skip_link_reset)
 {
 	azx_stop_chip(chip);
-	azx_enter_link_reset(chip);
+	if (!skip_link_reset)
+		azx_enter_link_reset(chip);
 	azx_clear_irq_pending(chip);
 	display_power(chip, false);
 }
@@ -895,6 +897,11 @@ static void azx_shutdown_chip(struct azx *chip)
 static DEFINE_MUTEX(card_list_lock);
 static LIST_HEAD(card_list);
 
+static void azx_shutdown_chip(struct azx *chip)
+{
+	__azx_shutdown_chip(chip, false);
+}
+
 static void azx_add_card_list(struct azx *chip)
 {
 	struct hda_intel *hda = container_of(chip, struct hda_intel, chip);
@@ -1717,7 +1724,7 @@ static void azx_check_snoop_available(struct azx *chip)
 
 static void azx_probe_work(struct work_struct *work)
 {
-	struct hda_intel *hda = container_of(work, struct hda_intel, probe_work);
+	struct hda_intel *hda = container_of(work, struct hda_intel, probe_work.work);
 	azx_probe_continue(&hda->chip);
 }
 
@@ -1822,7 +1829,7 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci,
 	}
 
 	/* continue probing in work context as may trigger request module */
-	INIT_WORK(&hda->probe_work, azx_probe_work);
+	INIT_DELAYED_WORK(&hda->probe_work, azx_probe_work);
 
 	*rchip = chip;
 
@@ -2136,7 +2143,7 @@ static int azx_probe(struct pci_dev *pci,
 #endif
 
 	if (schedule_probe)
-		schedule_work(&hda->probe_work);
+		schedule_delayed_work(&hda->probe_work, 0);
 
 	dev++;
 	if (chip->disabled)
@@ -2222,6 +2229,11 @@ static int azx_probe_continue(struct azx *chip)
 	int dev = chip->dev_index;
 	int err;
 
+	if (chip->disabled || hda->init_failed)
+		return -EIO;
+	if (hda->probe_retry)
+		goto probe_retry;
+
 	to_hda_bus(bus)->bus_probing = 1;
 	hda->probe_continued = 1;
 
@@ -2283,10 +2295,20 @@ static int azx_probe_continue(struct azx *chip)
 #endif
 	}
 #endif
+
+ probe_retry:
 	if (bus->codec_mask && !(probe_only[dev] & 1)) {
 		err = azx_codec_configure(chip);
-		if (err < 0)
+		if (err) {
+			if ((chip->driver_caps & AZX_DCAPS_RETRY_PROBE) &&
+			    ++hda->probe_retry < 60) {
+				schedule_delayed_work(&hda->probe_work,
+						      msecs_to_jiffies(1000));
+				return 0; /* keep things up */
+			}
+			dev_err(chip->card->dev, "Cannot probe codecs, giving up\n");
 			goto out_free;
+		}
 	}
 
 	err = snd_card_register(chip->card);
@@ -2316,6 +2338,7 @@ out_free:
 		display_power(chip, false);
 	complete_all(&hda->probe_wait);
 	to_hda_bus(bus)->bus_probing = 0;
+	hda->probe_retry = 0;
 	return 0;
 }
 
@@ -2341,7 +2364,7 @@ static void azx_remove(struct pci_dev *pci)
 		 * device during cancel_work_sync() call.
 		 */
 		device_unlock(&pci->dev);
-		cancel_work_sync(&hda->probe_work);
+		cancel_delayed_work_sync(&hda->probe_work);
 		device_lock(&pci->dev);
 
 		snd_card_free(card);
@@ -2357,7 +2380,7 @@ static void azx_shutdown(struct pci_dev *pci)
 		return;
 	chip = card->private_data;
 	if (chip && chip->running)
-		azx_shutdown_chip(chip);
+		__azx_shutdown_chip(chip, true);
 }
 
 /* PCI IDs */
diff --git a/sound/pci/hda/hda_intel.h b/sound/pci/hda/hda_intel.h
index 3fb119f09040..0f39418f9328 100644
--- a/sound/pci/hda/hda_intel.h
+++ b/sound/pci/hda/hda_intel.h
@@ -14,7 +14,7 @@ struct hda_intel {
 
 	/* sync probing */
 	struct completion probe_wait;
-	struct work_struct probe_work;
+	struct delayed_work probe_work;
 
 	/* card list (for power_save trigger) */
 	struct list_head list;
@@ -30,6 +30,8 @@ struct hda_intel {
 	unsigned int freed:1; /* resources already released */
 
 	bool need_i915_power:1; /* the hda controller needs i915 power */
+
+	int probe_retry;	/* being probe-retry */
 };
 
 #endif
diff --git a/sound/pci/hda/patch_cs8409.c b/sound/pci/hda/patch_cs8409.c
index 3c7ef55d016e..31ff11ab868e 100644
--- a/sound/pci/hda/patch_cs8409.c
+++ b/sound/pci/hda/patch_cs8409.c
@@ -1207,6 +1207,9 @@ void dolphin_fixups(struct hda_codec *codec, const struct hda_fixup *fix, int ac
 		snd_hda_jack_add_kctl(codec, DOLPHIN_LO_PIN_NID, "Line Out", true,
 				      SND_JACK_HEADPHONE, NULL);
 
+		snd_hda_jack_add_kctl(codec, DOLPHIN_AMIC_PIN_NID, "Microphone", true,
+				      SND_JACK_MICROPHONE, NULL);
+
 		cs8409_fix_caps(codec, DOLPHIN_HP_PIN_NID);
 		cs8409_fix_caps(codec, DOLPHIN_LO_PIN_NID);
 		cs8409_fix_caps(codec, DOLPHIN_AMIC_PIN_NID);
diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c
index 8b7a389b6aed..22d27b12c4e7 100644
--- a/sound/pci/hda/patch_realtek.c
+++ b/sound/pci/hda/patch_realtek.c
@@ -526,6 +526,8 @@ static void alc_shutup_pins(struct hda_codec *codec)
 	struct alc_spec *spec = codec->spec;
 
 	switch (codec->core.vendor_id) {
+	case 0x10ec0236:
+	case 0x10ec0256:
 	case 0x10ec0283:
 	case 0x10ec0286:
 	case 0x10ec0288:
@@ -2537,7 +2539,8 @@ static const struct snd_pci_quirk alc882_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1558, 0x67e1, "Clevo PB71[DE][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
 	SND_PCI_QUIRK(0x1558, 0x67e5, "Clevo PC70D[PRS](?:-D|-G)?", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
 	SND_PCI_QUIRK(0x1558, 0x70d1, "Clevo PC70[ER][CDF]", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
-	SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+	SND_PCI_QUIRK(0x1558, 0x7714, "Clevo X170SM", ALC1220_FIXUP_CLEVO_PB51ED_PINS),
+	SND_PCI_QUIRK(0x1558, 0x7715, "Clevo X170KM-G", ALC1220_FIXUP_CLEVO_PB51ED),
 	SND_PCI_QUIRK(0x1558, 0x9501, "Clevo P950HR", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1558, 0x9506, "Clevo P955HQ", ALC1220_FIXUP_CLEVO_P950),
 	SND_PCI_QUIRK(0x1558, 0x950a, "Clevo P955H[PR]", ALC1220_FIXUP_CLEVO_P950),
@@ -3528,7 +3531,8 @@ static void alc256_shutup(struct hda_codec *codec)
 	/* If disable 3k pulldown control for alc257, the Mic detection will not work correctly
 	 * when booting with headset plugged. So skip setting it for the codec alc257
 	 */
-	if (codec->core.vendor_id != 0x10ec0257)
+	if (spec->codec_variant != ALC269_TYPE_ALC257 &&
+	    spec->codec_variant != ALC269_TYPE_ALC256)
 		alc_update_coef_idx(codec, 0x46, 0, 3 << 12);
 
 	if (!spec->no_shutup_pins)
@@ -6429,12 +6433,44 @@ static void alc_fixup_thinkpad_acpi(struct hda_codec *codec,
 	hda_fixup_thinkpad_acpi(codec, fix, action);
 }
 
+/* Fixup for Lenovo Legion 15IMHg05 speaker output on headset removal. */
+static void alc287_fixup_legion_15imhg05_speakers(struct hda_codec *codec,
+						  const struct hda_fixup *fix,
+						  int action)
+{
+	struct alc_spec *spec = codec->spec;
+
+	switch (action) {
+	case HDA_FIXUP_ACT_PRE_PROBE:
+		spec->gen.suppress_auto_mute = 1;
+		break;
+	}
+}
+
 /* for alc295_fixup_hp_top_speakers */
 #include "hp_x360_helper.c"
 
 /* for alc285_fixup_ideapad_s740_coef() */
 #include "ideapad_s740_helper.c"
 
+static void alc256_fixup_tongfang_reset_persistent_settings(struct hda_codec *codec,
+							    const struct hda_fixup *fix,
+							    int action)
+{
+	/*
+	* A certain other OS sets these coeffs to different values. On at least one TongFang
+	* barebone these settings might survive even a cold reboot. So to restore a clean slate the
+	* values are explicitly reset to default here. Without this, the external microphone is
+	* always in a plugged-in state, while the internal microphone is always in an unplugged
+	* state, breaking the ability to use the internal microphone.
+	*/
+	alc_write_coef_idx(codec, 0x24, 0x0000);
+	alc_write_coef_idx(codec, 0x26, 0x0000);
+	alc_write_coef_idx(codec, 0x29, 0x3000);
+	alc_write_coef_idx(codec, 0x37, 0xfe05);
+	alc_write_coef_idx(codec, 0x45, 0x5089);
+}
+
 enum {
 	ALC269_FIXUP_GPIO2,
 	ALC269_FIXUP_SONY_VAIO,
@@ -6646,6 +6682,11 @@ enum {
 	ALC623_FIXUP_LENOVO_THINKSTATION_P340,
 	ALC255_FIXUP_ACER_HEADPHONE_AND_MIC,
 	ALC236_FIXUP_HP_LIMIT_INT_MIC_BOOST,
+	ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS,
+	ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE,
+	ALC287_FIXUP_YOGA7_14ITL_SPEAKERS,
+	ALC287_FIXUP_13S_GEN2_SPEAKERS,
+	ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS,
 };
 
 static const struct hda_fixup alc269_fixups[] = {
@@ -8236,6 +8277,117 @@ static const struct hda_fixup alc269_fixups[] = {
 		.chained = true,
 		.chain_id = ALC236_FIXUP_HP_MUTE_LED_MICMUTE_VREF,
 	},
+	[ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS] = {
+		.type = HDA_FIXUP_VERBS,
+		//.v.verbs = legion_15imhg05_coefs,
+		.v.verbs = (const struct hda_verb[]) {
+			 // set left speaker Legion 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x1a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 // set right speaker Legion 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x42 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			 {}
+		},
+		.chained = true,
+		.chain_id = ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE,
+	},
+	[ALC287_FIXUP_LEGION_15IMHG05_AUTOMUTE] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc287_fixup_legion_15imhg05_speakers,
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE,
+	},
+	[ALC287_FIXUP_YOGA7_14ITL_SPEAKERS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			 // set left speaker Yoga 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x1a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 // set right speaker Yoga 7i.
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x46 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xc },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2a },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+
+			 { 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			 { 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			 {}
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE,
+	},
+	[ALC287_FIXUP_13S_GEN2_SPEAKERS] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x41 },
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x24 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x42 },
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x26 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x2 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0xb020 },
+			{}
+		},
+		.chained = true,
+		.chain_id = ALC269_FIXUP_HEADSET_MODE,
+	},
+	[ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc256_fixup_tongfang_reset_persistent_settings,
+	},
 };
 
 static const struct snd_pci_quirk alc269_fixup_tbl[] = {
@@ -8327,6 +8479,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK),
+	SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK),
+	SND_PCI_QUIRK(0x1028, 0x0a9d, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
+	SND_PCI_QUIRK(0x1028, 0x0a9e, "Dell Latitude 5430", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2),
@@ -8630,6 +8785,10 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME),
 	SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF),
 	SND_PCI_QUIRK(0x17aa, 0x3843, "Yoga 9i", ALC287_FIXUP_IDEAPAD_BASS_SPK_AMP),
+	SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS),
+	SND_PCI_QUIRK(0x17aa, 0x3852, "Lenovo Yoga 7 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
+	SND_PCI_QUIRK(0x17aa, 0x3853, "Lenovo Yoga 7 15ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS),
+	SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS),
 	SND_PCI_QUIRK(0x17aa, 0x3902, "Lenovo E50-80", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
 	SND_PCI_QUIRK(0x17aa, 0x3977, "IdeaPad S210", ALC283_FIXUP_INT_MIC),
 	SND_PCI_QUIRK(0x17aa, 0x3978, "Lenovo B50-70", ALC269_FIXUP_DMIC_THINKPAD_ACPI),
@@ -8660,6 +8819,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1b7d, 0xa831, "Ordissimo EVE2 ", ALC269VB_FIXUP_ORDISSIMO_EVE2), /* Also known as Malata PC-B1303 */
 	SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802),
 	SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X),
+	SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_TONGFANG_RESET_PERSISTENT_SETTINGS),
 	SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE),
 	SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC),
@@ -10037,6 +10197,9 @@ enum {
 	ALC671_FIXUP_HP_HEADSET_MIC2,
 	ALC662_FIXUP_ACER_X2660G_HEADSET_MODE,
 	ALC662_FIXUP_ACER_NITRO_HEADSET_MODE,
+	ALC668_FIXUP_ASUS_NO_HEADSET_MIC,
+	ALC668_FIXUP_HEADSET_MIC,
+	ALC668_FIXUP_MIC_DET_COEF,
 };
 
 static const struct hda_fixup alc662_fixups[] = {
@@ -10420,6 +10583,29 @@ static const struct hda_fixup alc662_fixups[] = {
 		.chained = true,
 		.chain_id = ALC662_FIXUP_USI_FUNC
 	},
+	[ALC668_FIXUP_ASUS_NO_HEADSET_MIC] = {
+		.type = HDA_FIXUP_PINS,
+		.v.pins = (const struct hda_pintbl[]) {
+			{ 0x1b, 0x04a1112c },
+			{ }
+		},
+		.chained = true,
+		.chain_id = ALC668_FIXUP_HEADSET_MIC
+	},
+	[ALC668_FIXUP_HEADSET_MIC] = {
+		.type = HDA_FIXUP_FUNC,
+		.v.func = alc269_fixup_headset_mic,
+		.chained = true,
+		.chain_id = ALC668_FIXUP_MIC_DET_COEF
+	},
+	[ALC668_FIXUP_MIC_DET_COEF] = {
+		.type = HDA_FIXUP_VERBS,
+		.v.verbs = (const struct hda_verb[]) {
+			{ 0x20, AC_VERB_SET_COEF_INDEX, 0x15 },
+			{ 0x20, AC_VERB_SET_PROC_COEF, 0x0d60 },
+			{}
+		},
+	},
 };
 
 static const struct snd_pci_quirk alc662_fixup_tbl[] = {
@@ -10455,6 +10641,7 @@ static const struct snd_pci_quirk alc662_fixup_tbl[] = {
 	SND_PCI_QUIRK(0x1043, 0x15a7, "ASUS UX51VZH", ALC662_FIXUP_BASS_16),
 	SND_PCI_QUIRK(0x1043, 0x177d, "ASUS N551", ALC668_FIXUP_ASUS_Nx51),
 	SND_PCI_QUIRK(0x1043, 0x17bd, "ASUS N751", ALC668_FIXUP_ASUS_Nx51),
+	SND_PCI_QUIRK(0x1043, 0x185d, "ASUS G551JW", ALC668_FIXUP_ASUS_NO_HEADSET_MIC),
 	SND_PCI_QUIRK(0x1043, 0x1963, "ASUS X71SL", ALC662_FIXUP_ASUS_MODE8),
 	SND_PCI_QUIRK(0x1043, 0x1b73, "ASUS N55SF", ALC662_FIXUP_BASS_16),
 	SND_PCI_QUIRK(0x1043, 0x1bf3, "ASUS N76VZ", ALC662_FIXUP_BASS_MODE4_CHMAP),
diff --git a/sound/pci/pcxhr/pcxhr_core.c b/sound/pci/pcxhr/pcxhr_core.c
index 87d24224c042..23f253effb4f 100644
--- a/sound/pci/pcxhr/pcxhr_core.c
+++ b/sound/pci/pcxhr/pcxhr_core.c
@@ -52,7 +52,7 @@
 #define PCXHR_DSP 2
 
 #if (PCXHR_DSP_OFFSET_MAX > PCXHR_PLX_OFFSET_MIN)
-#undef  PCXHR_REG_TO_PORT(x)
+#error  PCXHR_REG_TO_PORT(x)
 #else
 #define PCXHR_REG_TO_PORT(x)	((x)>PCXHR_DSP_OFFSET_MAX ? PCXHR_PLX : PCXHR_DSP)
 #endif
diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c
index a961f837cd09..bda66b30e063 100644
--- a/sound/soc/fsl/fsl_esai.c
+++ b/sound/soc/fsl/fsl_esai.c
@@ -1073,6 +1073,16 @@ static int fsl_esai_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto err_pm_get_sync;
 
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
+		goto err_pm_get_sync;
+	}
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component,
 					      &fsl_esai_dai, 1);
 	if (ret) {
@@ -1082,12 +1092,6 @@ static int fsl_esai_probe(struct platform_device *pdev)
 
 	INIT_WORK(&esai_priv->work, fsl_esai_hw_reset);
 
-	ret = imx_pcm_dma_init(pdev, IMX_ESAI_DMABUF_SIZE);
-	if (ret) {
-		dev_err(&pdev->dev, "failed to init imx pcm dma: %d\n", ret);
-		goto err_pm_get_sync;
-	}
-
 	return ret;
 
 err_pm_get_sync:
diff --git a/sound/soc/fsl/fsl_micfil.c b/sound/soc/fsl/fsl_micfil.c
index 8c0c75ce9490..9f90989ac59a 100644
--- a/sound/soc/fsl/fsl_micfil.c
+++ b/sound/soc/fsl/fsl_micfil.c
@@ -737,18 +737,23 @@ static int fsl_micfil_probe(struct platform_device *pdev)
 	pm_runtime_enable(&pdev->dev);
 	regcache_cache_only(micfil->regmap, true);
 
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to pcm register\n");
+		return ret;
+	}
+
 	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_micfil_component,
 					      &fsl_micfil_dai, 1);
 	if (ret) {
 		dev_err(&pdev->dev, "failed to register component %s\n",
 			fsl_micfil_component.name);
-		return ret;
 	}
 
-	ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0);
-	if (ret)
-		dev_err(&pdev->dev, "failed to pcm register\n");
-
 	return ret;
 }
 
diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c
index 223fcd15bfcc..38f6362099d5 100644
--- a/sound/soc/fsl/fsl_sai.c
+++ b/sound/soc/fsl/fsl_sai.c
@@ -1152,11 +1152,10 @@ static int fsl_sai_probe(struct platform_device *pdev)
 	if (ret < 0)
 		goto err_pm_get_sync;
 
-	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
-					      &sai->cpu_dai_drv, 1);
-	if (ret)
-		goto err_pm_get_sync;
-
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
 	if (sai->soc_data->use_imx_pcm) {
 		ret = imx_pcm_dma_init(pdev, IMX_SAI_DMABUF_SIZE);
 		if (ret)
@@ -1167,6 +1166,11 @@ static int fsl_sai_probe(struct platform_device *pdev)
 			goto err_pm_get_sync;
 	}
 
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_component,
+					      &sai->cpu_dai_drv, 1);
+	if (ret)
+		goto err_pm_get_sync;
+
 	return ret;
 
 err_pm_get_sync:
diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c
index 8ffb1a6048d6..1c53719bb61e 100644
--- a/sound/soc/fsl/fsl_spdif.c
+++ b/sound/soc/fsl/fsl_spdif.c
@@ -1434,16 +1434,20 @@ static int fsl_spdif_probe(struct platform_device *pdev)
 	pm_runtime_enable(&pdev->dev);
 	regcache_cache_only(spdif_priv->regmap, true);
 
-	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
-					      &spdif_priv->cpu_dai_drv, 1);
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE);
 	if (ret) {
-		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
+		dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n");
 		goto err_pm_disable;
 	}
 
-	ret = imx_pcm_dma_init(pdev, IMX_SPDIF_DMABUF_SIZE);
+	ret = devm_snd_soc_register_component(&pdev->dev, &fsl_spdif_component,
+					      &spdif_priv->cpu_dai_drv, 1);
 	if (ret) {
-		dev_err_probe(&pdev->dev, ret, "imx_pcm_dma_init failed\n");
+		dev_err(&pdev->dev, "failed to register DAI: %d\n", ret);
 		goto err_pm_disable;
 	}
 
diff --git a/sound/soc/fsl/fsl_xcvr.c b/sound/soc/fsl/fsl_xcvr.c
index 31c5ee641fe7..7ba2fd15132d 100644
--- a/sound/soc/fsl/fsl_xcvr.c
+++ b/sound/soc/fsl/fsl_xcvr.c
@@ -1215,18 +1215,23 @@ static int fsl_xcvr_probe(struct platform_device *pdev)
 	pm_runtime_enable(dev);
 	regcache_cache_only(xcvr->regmap, true);
 
+	/*
+	 * Register platform component before registering cpu dai for there
+	 * is not defer probe for platform component in snd_soc_add_pcm_runtime().
+	 */
+	ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0);
+	if (ret) {
+		dev_err(dev, "failed to pcm register\n");
+		return ret;
+	}
+
 	ret = devm_snd_soc_register_component(dev, &fsl_xcvr_comp,
 					      &fsl_xcvr_dai, 1);
 	if (ret) {
 		dev_err(dev, "failed to register component %s\n",
 			fsl_xcvr_comp.name);
-		return ret;
 	}
 
-	ret = devm_snd_dmaengine_pcm_register(dev, NULL, 0);
-	if (ret)
-		dev_err(dev, "failed to pcm register\n");
-
 	return ret;
 }
 
diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c
index 6602eda89e8e..6b06248a9327 100644
--- a/sound/soc/intel/boards/sof_sdw.c
+++ b/sound/soc/intel/boards/sof_sdw.c
@@ -929,6 +929,11 @@ static int create_sdw_dailink(struct snd_soc_card *card,
 			      cpus + *cpu_id, cpu_dai_num,
 			      codecs, codec_num,
 			      NULL, &sdw_ops);
+		/*
+		 * SoundWire DAILINKs use 'stream' functions and Bank Switch operations
+		 * based on wait_for_completion(), tag them as 'nonatomic'.
+		 */
+		dai_links[*be_index].nonatomic = true;
 
 		ret = set_codec_init_func(card, link, dai_links + (*be_index)++,
 					  playback, group_id);
diff --git a/sound/soc/mediatek/Kconfig b/sound/soc/mediatek/Kconfig
index 5a2f4667d50b..81ad2dcee9eb 100644
--- a/sound/soc/mediatek/Kconfig
+++ b/sound/soc/mediatek/Kconfig
@@ -1,6 +1,7 @@
 # SPDX-License-Identifier: GPL-2.0-only
 config SND_SOC_MEDIATEK
 	tristate
+	select REGMAP_MMIO
 
 config SND_SOC_MT2701
 	tristate "ASoC support for Mediatek MT2701 chip"
@@ -188,7 +189,9 @@ config SND_SOC_MT8192_MT6359_RT1015_RT5682
 config SND_SOC_MT8195
 	tristate "ASoC support for Mediatek MT8195 chip"
 	depends on ARCH_MEDIATEK || COMPILE_TEST
+	depends on COMMON_CLK
 	select SND_SOC_MEDIATEK
+	select MFD_SYSCON if SND_SOC_MT6359
 	help
 	  This adds ASoC platform driver support for Mediatek MT8195 chip
 	  that can be used with other codecs.
diff --git a/sound/soc/mediatek/common/mtk-afe-fe-dai.c b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
index baaa5881b1d4..e95c7c018e7d 100644
--- a/sound/soc/mediatek/common/mtk-afe-fe-dai.c
+++ b/sound/soc/mediatek/common/mtk-afe-fe-dai.c
@@ -334,9 +334,11 @@ int mtk_afe_suspend(struct snd_soc_component *component)
 			devm_kcalloc(dev, afe->reg_back_up_list_num,
 				     sizeof(unsigned int), GFP_KERNEL);
 
-	for (i = 0; i < afe->reg_back_up_list_num; i++)
-		regmap_read(regmap, afe->reg_back_up_list[i],
-			    &afe->reg_back_up[i]);
+	if (afe->reg_back_up) {
+		for (i = 0; i < afe->reg_back_up_list_num; i++)
+			regmap_read(regmap, afe->reg_back_up_list[i],
+				    &afe->reg_back_up[i]);
+	}
 
 	afe->suspended = true;
 	afe->runtime_suspend(dev);
@@ -356,12 +358,13 @@ int mtk_afe_resume(struct snd_soc_component *component)
 
 	afe->runtime_resume(dev);
 
-	if (!afe->reg_back_up)
+	if (!afe->reg_back_up) {
 		dev_dbg(dev, "%s no reg_backup\n", __func__);
-
-	for (i = 0; i < afe->reg_back_up_list_num; i++)
-		mtk_regmap_write(regmap, afe->reg_back_up_list[i],
-				 afe->reg_back_up[i]);
+	} else {
+		for (i = 0; i < afe->reg_back_up_list_num; i++)
+			mtk_regmap_write(regmap, afe->reg_back_up_list[i],
+					 afe->reg_back_up[i]);
+	}
 
 	afe->suspended = false;
 	return 0;
diff --git a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c
index c97ace7387b4..de09f67c0450 100644
--- a/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c
+++ b/sound/soc/mediatek/mt8195/mt8195-mt6359-rt1019-rt5682.c
@@ -424,8 +424,8 @@ static int mt8195_hdmi_codec_init(struct snd_soc_pcm_runtime *rtd)
 	return snd_soc_component_set_jack(cmpnt_codec, &priv->hdmi_jack, NULL);
 }
 
-static int mt8195_hdmitx_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
-					      struct snd_pcm_hw_params *params)
+static int mt8195_dptx_hw_params_fixup(struct snd_soc_pcm_runtime *rtd,
+				       struct snd_pcm_hw_params *params)
 
 {
 	/* fix BE i2s format to 32bit, clean param mask first */
@@ -902,7 +902,7 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = {
 		.no_pcm = 1,
 		.dpcm_playback = 1,
 		.ops = &mt8195_dptx_ops,
-		.be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup,
+		.be_hw_params_fixup = mt8195_dptx_hw_params_fixup,
 		SND_SOC_DAILINK_REG(DPTX_BE),
 	},
 	[DAI_LINK_ETDM1_IN_BE] = {
@@ -953,7 +953,6 @@ static struct snd_soc_dai_link mt8195_mt6359_rt1019_rt5682_dai_links[] = {
 			SND_SOC_DAIFMT_NB_NF |
 			SND_SOC_DAIFMT_CBS_CFS,
 		.dpcm_playback = 1,
-		.be_hw_params_fixup = mt8195_hdmitx_dptx_hw_params_fixup,
 		SND_SOC_DAILINK_REG(ETDM3_OUT_BE),
 	},
 	[DAI_LINK_PCM1_BE] = {
diff --git a/sound/soc/sof/core.c b/sound/soc/sof/core.c
index 3e4dd4a86363..59d0d7b2b55c 100644
--- a/sound/soc/sof/core.c
+++ b/sound/soc/sof/core.c
@@ -371,7 +371,6 @@ int snd_sof_device_remove(struct device *dev)
 			dev_warn(dev, "error: %d failed to prepare DSP for device removal",
 				 ret);
 
-		snd_sof_fw_unload(sdev);
 		snd_sof_ipc_free(sdev);
 		snd_sof_free_debug(sdev);
 		snd_sof_free_trace(sdev);
@@ -394,8 +393,7 @@ int snd_sof_device_remove(struct device *dev)
 		snd_sof_remove(sdev);
 
 	/* release firmware */
-	release_firmware(pdata->fw);
-	pdata->fw = NULL;
+	snd_sof_fw_unload(sdev);
 
 	return 0;
 }
diff --git a/sound/soc/sof/imx/imx8.c b/sound/soc/sof/imx/imx8.c
index 12fedf0984bd..7e9723a10d02 100644
--- a/sound/soc/sof/imx/imx8.c
+++ b/sound/soc/sof/imx/imx8.c
@@ -365,7 +365,14 @@ static int imx8_remove(struct snd_sof_dev *sdev)
 /* on i.MX8 there is 1 to 1 match between type and BAR idx */
 static int imx8_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 {
-	return type;
+	/* Only IRAM and SRAM bars are valid */
+	switch (type) {
+	case SOF_FW_BLK_TYPE_IRAM:
+	case SOF_FW_BLK_TYPE_SRAM:
+		return type;
+	default:
+		return -EINVAL;
+	}
 }
 
 static void imx8_ipc_msg_data(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/imx/imx8m.c b/sound/soc/sof/imx/imx8m.c
index cb822d953767..892e1482f97f 100644
--- a/sound/soc/sof/imx/imx8m.c
+++ b/sound/soc/sof/imx/imx8m.c
@@ -228,7 +228,14 @@ static int imx8m_remove(struct snd_sof_dev *sdev)
 /* on i.MX8 there is 1 to 1 match between type and BAR idx */
 static int imx8m_get_bar_index(struct snd_sof_dev *sdev, u32 type)
 {
-	return type;
+	/* Only IRAM and SRAM bars are valid */
+	switch (type) {
+	case SOF_FW_BLK_TYPE_IRAM:
+	case SOF_FW_BLK_TYPE_SRAM:
+		return type;
+	default:
+		return -EINVAL;
+	}
 }
 
 static void imx8m_ipc_msg_data(struct snd_sof_dev *sdev,
diff --git a/sound/soc/sof/loader.c b/sound/soc/sof/loader.c
index 2b38a77cd594..bb79c77775b3 100644
--- a/sound/soc/sof/loader.c
+++ b/sound/soc/sof/loader.c
@@ -729,10 +729,10 @@ int snd_sof_load_firmware_raw(struct snd_sof_dev *sdev)
 	ret = request_firmware(&plat_data->fw, fw_filename, sdev->dev);
 
 	if (ret < 0) {
-		dev_err(sdev->dev, "error: request firmware %s failed err: %d\n",
-			fw_filename, ret);
 		dev_err(sdev->dev,
-			"you may need to download the firmware from https://github.com/thesofproject/sof-bin/\n");
+			"error: sof firmware file is missing, you might need to\n");
+		dev_err(sdev->dev,
+			"       download it from https://github.com/thesofproject/sof-bin/\n");
 		goto err;
 	} else {
 		dev_dbg(sdev->dev, "request_firmware %s successful\n",
@@ -880,5 +880,7 @@ EXPORT_SYMBOL(snd_sof_run_firmware);
 void snd_sof_fw_unload(struct snd_sof_dev *sdev)
 {
 	/* TODO: support module unloading at runtime */
+	release_firmware(sdev->pdata->fw);
+	sdev->pdata->fw = NULL;
 }
 EXPORT_SYMBOL(snd_sof_fw_unload);
diff --git a/sound/soc/sof/trace.c b/sound/soc/sof/trace.c
index f72a6e83e6af..58f6ca5cf491 100644
--- a/sound/soc/sof/trace.c
+++ b/sound/soc/sof/trace.c
@@ -530,7 +530,6 @@ void snd_sof_trace_notify_for_error(struct snd_sof_dev *sdev)
 		return;
 
 	if (sdev->dtrace_is_enabled) {
-		dev_err(sdev->dev, "error: waking up any trace sleepers\n");
 		sdev->dtrace_error = true;
 		wake_up(&sdev->trace_sleep);
 	}
diff --git a/sound/soc/sof/xtensa/core.c b/sound/soc/sof/xtensa/core.c
index bbb9a2282ed9..f6e3411b33cf 100644
--- a/sound/soc/sof/xtensa/core.c
+++ b/sound/soc/sof/xtensa/core.c
@@ -122,9 +122,9 @@ static void xtensa_stack(struct snd_sof_dev *sdev, void *oops, u32 *stack,
 	 * 0x0049fbb0: 8000f2d0 0049fc00 6f6c6c61 00632e63
 	 */
 	for (i = 0; i < stack_words; i += 4) {
-		hex_dump_to_buffer(stack + i * 4, 16, 16, 4,
+		hex_dump_to_buffer(stack + i, 16, 16, 4,
 				   buf, sizeof(buf), false);
-		dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i, buf);
+		dev_err(sdev->dev, "0x%08x: %s\n", stack_ptr + i * 4, buf);
 	}
 }
 
diff --git a/sound/usb/card.c b/sound/usb/card.c
index fd570a42f043..1764b9302d46 100644
--- a/sound/usb/card.c
+++ b/sound/usb/card.c
@@ -1054,7 +1054,7 @@ static int usb_audio_suspend(struct usb_interface *intf, pm_message_t message)
 	return 0;
 }
 
-static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
+static int usb_audio_resume(struct usb_interface *intf)
 {
 	struct snd_usb_audio *chip = usb_get_intfdata(intf);
 	struct snd_usb_stream *as;
@@ -1080,7 +1080,7 @@ static int __usb_audio_resume(struct usb_interface *intf, bool reset_resume)
 	 * we just notify and restart the mixers
 	 */
 	list_for_each_entry(mixer, &chip->mixer_list, list) {
-		err = snd_usb_mixer_resume(mixer, reset_resume);
+		err = snd_usb_mixer_resume(mixer);
 		if (err < 0)
 			goto err_out;
 	}
@@ -1100,20 +1100,10 @@ err_out:
 	atomic_dec(&chip->active); /* allow autopm after this point */
 	return err;
 }
-
-static int usb_audio_resume(struct usb_interface *intf)
-{
-	return __usb_audio_resume(intf, false);
-}
-
-static int usb_audio_reset_resume(struct usb_interface *intf)
-{
-	return __usb_audio_resume(intf, true);
-}
 #else
 #define usb_audio_suspend	NULL
 #define usb_audio_resume	NULL
-#define usb_audio_reset_resume	NULL
+#define usb_audio_resume	NULL
 #endif		/* CONFIG_PM */
 
 static const struct usb_device_id usb_audio_ids [] = {
@@ -1135,7 +1125,7 @@ static struct usb_driver usb_audio_driver = {
 	.disconnect =	usb_audio_disconnect,
 	.suspend =	usb_audio_suspend,
 	.resume =	usb_audio_resume,
-	.reset_resume =	usb_audio_reset_resume,
+	.reset_resume =	usb_audio_resume,
 	.id_table =	usb_audio_ids,
 	.supports_autosuspend = 1,
 };
diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c
index 43bc59575a6e..a2ce535df14b 100644
--- a/sound/usb/mixer.c
+++ b/sound/usb/mixer.c
@@ -3653,33 +3653,16 @@ static int restore_mixer_value(struct usb_mixer_elem_list *list)
 	return 0;
 }
 
-static int default_mixer_reset_resume(struct usb_mixer_elem_list *list)
-{
-	int err;
-
-	if (list->resume) {
-		err = list->resume(list);
-		if (err < 0)
-			return err;
-	}
-	return restore_mixer_value(list);
-}
-
-int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume)
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer)
 {
 	struct usb_mixer_elem_list *list;
-	usb_mixer_elem_resume_func_t f;
 	int id, err;
 
 	/* restore cached mixer values */
 	for (id = 0; id < MAX_ID_ELEMS; id++) {
 		for_each_mixer_elem(list, mixer, id) {
-			if (reset_resume)
-				f = list->reset_resume;
-			else
-				f = list->resume;
-			if (f) {
-				err = f(list);
+			if (list->resume) {
+				err = list->resume(list);
 				if (err < 0)
 					return err;
 			}
@@ -3700,7 +3683,6 @@ void snd_usb_mixer_elem_init_std(struct usb_mixer_elem_list *list,
 	list->id = unitid;
 	list->dump = snd_usb_mixer_dump_cval;
 #ifdef CONFIG_PM
-	list->resume = NULL;
-	list->reset_resume = default_mixer_reset_resume;
+	list->resume = restore_mixer_value;
 #endif
 }
diff --git a/sound/usb/mixer.h b/sound/usb/mixer.h
index 876bbc9a71ad..98ea24d91d80 100644
--- a/sound/usb/mixer.h
+++ b/sound/usb/mixer.h
@@ -70,7 +70,6 @@ struct usb_mixer_elem_list {
 	bool is_std_info;
 	usb_mixer_elem_dump_func_t dump;
 	usb_mixer_elem_resume_func_t resume;
-	usb_mixer_elem_resume_func_t reset_resume;
 };
 
 /* iterate over mixer element list of the given unit id */
@@ -121,7 +120,7 @@ int snd_usb_mixer_vol_tlv(struct snd_kcontrol *kcontrol, int op_flag,
 
 #ifdef CONFIG_PM
 int snd_usb_mixer_suspend(struct usb_mixer_interface *mixer);
-int snd_usb_mixer_resume(struct usb_mixer_interface *mixer, bool reset_resume);
+int snd_usb_mixer_resume(struct usb_mixer_interface *mixer);
 #endif
 
 int snd_usb_set_cur_mix_value(struct usb_mixer_elem_info *cval, int channel,
diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c
index a66ce0375fd9..46082dc57be0 100644
--- a/sound/usb/mixer_quirks.c
+++ b/sound/usb/mixer_quirks.c
@@ -151,7 +151,7 @@ static int add_single_ctl_with_resume(struct usb_mixer_interface *mixer,
 		*listp = list;
 	list->mixer = mixer;
 	list->id = id;
-	list->reset_resume = resume;
+	list->resume = resume;
 	kctl = snd_ctl_new1(knew, list);
 	if (!kctl) {
 		kfree(list);
diff --git a/sound/usb/mixer_scarlett_gen2.c b/sound/usb/mixer_scarlett_gen2.c
index 3d5848d5481b..53ebabf42472 100644
--- a/sound/usb/mixer_scarlett_gen2.c
+++ b/sound/usb/mixer_scarlett_gen2.c
@@ -2450,6 +2450,8 @@ static int scarlett2_update_monitor_other(struct usb_mixer_interface *mixer)
 		err = scarlett2_usb_get_config(mixer,
 					       SCARLETT2_CONFIG_TALKBACK_MAP,
 					       1, &bitmap);
+		if (err < 0)
+			return err;
 		for (i = 0; i < num_mixes; i++, bitmap >>= 1)
 			private->talkback_map[i] = bitmap & 1;
 	}
diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h
index e03043f7dad3..de18fff69280 100644
--- a/sound/usb/quirks-table.h
+++ b/sound/usb/quirks-table.h
@@ -78,6 +78,48 @@
 { USB_DEVICE_VENDOR_SPEC(0x041e, 0x3f19) },
 
 /*
+ * Creative Technology, Ltd Live! Cam Sync HD [VF0770]
+ * The device advertises 8 formats, but only a rate of 48kHz is honored by the
+ * hardware and 24 bits give chopped audio, so only report the one working
+ * combination.
+ */
+{
+	USB_DEVICE(0x041e, 0x4095),
+	.driver_info = (unsigned long) &(const struct snd_usb_audio_quirk) {
+		.ifnum = QUIRK_ANY_INTERFACE,
+		.type = QUIRK_COMPOSITE,
+		.data = &(const struct snd_usb_audio_quirk[]) {
+			{
+				.ifnum = 2,
+				.type = QUIRK_AUDIO_STANDARD_MIXER,
+			},
+			{
+				.ifnum = 3,
+				.type = QUIRK_AUDIO_FIXED_ENDPOINT,
+				.data = &(const struct audioformat) {
+					.formats = SNDRV_PCM_FMTBIT_S16_LE,
+					.channels = 2,
+					.fmt_bits = 16,
+					.iface = 3,
+					.altsetting = 4,
+					.altset_idx = 4,
+					.endpoint = 0x82,
+					.ep_attr = 0x05,
+					.rates = SNDRV_PCM_RATE_48000,
+					.rate_min = 48000,
+					.rate_max = 48000,
+					.nr_rates = 1,
+					.rate_table = (unsigned int[]) { 48000 },
+				},
+			},
+			{
+				.ifnum = -1
+			},
+		},
+	},
+},
+
+/*
  * HP Wireless Audio
  * When not ignored, causes instability issues for some users, forcing them to
  * skip the entire module.
diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c
index 6ee6d24c847f..889c855addfc 100644
--- a/sound/usb/quirks.c
+++ b/sound/usb/quirks.c
@@ -1900,6 +1900,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = {
 		   QUIRK_FLAG_CTL_MSG_DELAY | QUIRK_FLAG_IFACE_DELAY),
 	VENDOR_FLG(0x07fd, /* MOTU */
 		   QUIRK_FLAG_VALIDATE_RATES),
+	VENDOR_FLG(0x1235, /* Focusrite Novation */
+		   QUIRK_FLAG_VALIDATE_RATES),
 	VENDOR_FLG(0x152a, /* Thesycon devices */
 		   QUIRK_FLAG_DSD_RAW),
 	VENDOR_FLG(0x1de7, /* Phoenix Audio */